Sync when compressing HD FLV and F4V in Premiere Pro CS4

Question

Hi.  I'm trying to create PSAs for You Tube that were originally created with a small camera that produced HD Quicktime files.  For YouTube they don't have to be perfect but that is what I'm working with.  My computer only has 4gb of ram so I can't watch the video to test for syncronization since it just gives a slide show while editing.  It never does anything but a slide show with HD video.
When I watch the original unedited Quicktime file the sync was way off.  When I render out the original file in an FLV and F4V format just to test the sync it is PERFECT.  Then I edited the first video taken from that session and used FLV and the sync was PERFECT.  So I'm doing the second video, tested sync on the original footage, and its PERFECT.
So I thought it was okay to keep going.  Everything was perfect after Premiere rendered the raw footage. The first video worked, right? The second video's raw footage sync was fixed, right?
Now that I spent all the time to edit the 2nd video, sync is TERRIBLE.  I have no idea why.  F4V was worse than FLV, but neither was great for video 2.  So I guess I have two questions:
1) What format can I use to upload to YouTube that won't give me an unusable sync problem
2) Is there something I can do to see continuous synchronization within Premiere Pro CS4 without updating my hardware?  I am using Vista 64 bit, 4gb ram, dual core intel.  Do I have to make a major hardware upgrade just to get HD video not to be a slide show while running Premiere?
Thanks!

Answer 1

Dave,
Deleted all stuff...
You might try this "TEST" at some point if you're using equipment to record sound as outlined in the following article....at any rate you might find the article interesting anyway for the future...
----------------------sorry I should give credit to who wrote it but you can google the first line and find it maybe -----------
sync demuxed sound   The Problem -
Miniature audio recorders, be they minidisc or solid state, offer a very attractive, cost effective, alternative to the traditional use of radio microphones, for the capture of remote audio sources in video recordings
These machines record the audio in a variety of different formats - MP3, WMA, WAV, ATRAC -- and in a variety of qualities - but they all suffer from the same problem. It can be difficult to synchronise the remotely recorded audio, with the camera recorded audio.
Although the time honoured tradition of an audio 'clapper board' works well enough to align the beginning of the two soundtracks, there is a tendency for the two tracks to appear to 'drift' out of sync over time - and the longer the recording, the worse the problem. In fact they do not actually 'drift', but the gradual loss of sync is a function of the camera and audio recorder sample clocks being slightly different frequencies.
The simple reason for this loss of 'sync' therefore is that the audio is being recorded at different speeds. Although the recorder will specify the sampling frequency, the accuracy is then dependent on the internal crystal oscillator of the recorder - which is, of course, a different oscillator from the one controlling the camera timing. Although these crystals may typically be accurate to within 0.005% or better, even this degree of accuracy can cause the two recorded audio tracks to be out of sync by up to 300mS after 30 minutes. Even after 5 minutes, the 2 tracks can be some 60mS different in length, in a worst case scenario.
60mS difference will cause a distinctly audible 'echo' between the two audio tracks. (Even 20mS is detectable as a separate echo).
This difference is of course totally unacceptable, and corrections need to be applied. It is important to remember that even if your audio recorder crystal is absolutely accurate, and it is the camera clock that is slightly wrong, it is still the remote audio track that will have to be corrected. The 'wrong' camera audio track is the one in sync with the video, and has to be the 'master' track.
The Solution
It is important to realise that although the sample oscillators may vary with regard to each other, they tend to be remarkably accurate over time, referenced to themselves.
This means that if you know by how much the frequency your audio recorder sample 'clock' differs from the camera 'clock', then a single correction factor will bring both tracks back into sync.
A simple procedure is outlined below, using free programs, to enable you to calculate the correction factor required for your particular recorder.
Programs required (all free):
1) AoA audio Extractor --- (not necessary if you can extract the audio track from a video recording in your NLE)
2) Audacity , a free audio editor
3) Switch , a free audio file type converter
Step 1: Set up your camera and audio recorder next to each other, and start both running
Step 2: Make a good audio 'clapperboard' point by tapping a pencil firmly onto a hard surface, as close to both camera and recorder mics as possible. This will cause a fast 'edge' to become the reference point, which will be useful later.
Step3: Let both devices run for at least 10 minutes. 20 minutes are better, but not essential.
At the end of the period, (timing not critical), make a second similar 'clapperboard' reference point. Allow a few more seconds recording, and then stop both devices.
Step 4: Download the video from the camera, and the audio from the recorder to your computer.
Step 5: Open the video file in 'AoA audio Extractor', and extract the audio from the video. Convert this audio file (if necessary) to a 48KHz 16 bit LPCM (wav) file, using the 'Switch' program linked to above. This file will be your reference audio file.
Step 6: Open the WAV file saved in the previous step in 'Audacity' - a free audio editor.
It is now necessary to save only the part of the file between the two 'clapperboard' references, which hopefully stand out clearly (if you hit the pencil hard enough in steps 2 and 3!).
This must be done accurately, to within 100th of a second, or better.
Click and drag the file to highlight the beginning section, including the first 'clapperboard' waveform. Use the 'fit selection to window' tool (immediately to the right of the 'zoom' tools) to magnify. Delete as much as possible before the 'clapper' reference. Repeat the operation until you can clearly identify the first distinct waveform of the 'clapper' reference waveform. Click, hold and place the cursor directly over the first full (clipped) waveform, where it crosses the centre line. Delete every thing to the left of that point, by clicking and dragging to highlight, and then deleting with the scissors tool.
Step 7: Repeat the above procedure for the end part of the waveform, again using the start of the second 'clapper' waveform as your reference cutting point. In this case, cut everything to the right of your selected reference point (not the left, as in the first case.)
Step 8: It is necessary to record the exact length of this new cropped waveform. This information is listed at the bottom of the window, but in a rather inappropriate format. Locate the toolbar at the top of the window, click 'Edit'-'select all' and then 'Effects'-'change tempo' . Note the length of the file in seconds in the right hand box (the left hand box is greyed out).
Record this figure. It is the reference file length
Step 9: It is necessary to convert the file recorded on the audio recorder into an identical format to the camera audio track, namely 48 KHz 16 bit PCM. If the track is not already in this format, open it in Switch, a free file transfer program, and select the output encoding as 48 KHz 16 bit, mono or stereo to fit your audio file. This program will then convert whatever format your input file is (MP3, WMA, 44.1 KHz PCM, etc) into the required 48 KHz, 16 bit format required.
Step 10: Repeat the procedures in steps 6 and 7 for this new file. Edit the 'clapper' points with as much accuracy as before, but notice that the 'clapper' waveform will appear different from the first, although hopefully with as clearly defined first ' zero crossing' point.
Step 11: Repeat the procedure in step 8, and you will notice that the overall file length is different.
Overwrite the original file length into the right hand box, and take careful note of the resulting figure in the percent change box. Make sure you record whether it is a plus or minus value - it could be either! This is the most important figure, because it represents the percentage variation between the two units.
Once you have obtained this value, simply apply it to all files recorded with the same recorder and this camera. Providing you can align the start of the file with a single 'clapperboard' mark, there is no need to repeat the rest. Simply change the length of the external audio file by the correct percentage value, to allow the two audio tracks to remain synchronised.
It may seem a long winded process, but it only needs to be done once, and could then save quite a lot of time, against the option of trying to align tracks manually in your NLE, in small sections.
You may of course wish to use an alternative audio editor, and the details of this procedure may then be slightly different, but the principle remains the same.